Unflaking PeerConnectionIntegrationTest.DtmfSenderObserver.

webrtc/pc/peerconnection_integrationtest.cc

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 940 matching lines @@
     }
     if (callee_) {
       callee_->set_signaling_message_receiver(nullptr);
     }
   }
 
   bool SignalingStateStable() {
     return caller_->SignalingStateStable() && callee_->SignalingStateStable();
   }
 
+  bool DtlsConnected() {
+    // TODO(deadbeef): kIceConnectionConnected currently means both ICE and DTLS
+    // are connected. This is an important distinction. Once we have separate
+    // ICE and DTLS state, this check needs to use the DTLS state.
+    return (callee()->ice_connection_state() ==
+                webrtc::PeerConnectionInterface::kIceConnectionConnected ||
+            callee()->ice_connection_state() ==
+                webrtc::PeerConnectionInterface::kIceConnectionCompleted) &&
+           (caller()->ice_connection_state() ==
+                webrtc::PeerConnectionInterface::kIceConnectionConnected ||
+            caller()->ice_connection_state() ==
+                webrtc::PeerConnectionInterface::kIceConnectionCompleted);
+  }
+
   bool CreatePeerConnectionWrappers() {
     return CreatePeerConnectionWrappersWithConfig(
         PeerConnectionInterface::RTCConfiguration(),
         PeerConnectionInterface::RTCConfiguration());
   }
 
   bool CreatePeerConnectionWrappersWithConstraints(
       MediaConstraintsInterface* caller_constraints,
       MediaConstraintsInterface* callee_constraints) {
     caller_.reset(PeerConnectionWrapper::CreateWithConstraints(
@@ skipping 267 matching lines @@
 // Verifies the DtmfSenderObserver callbacks for a DtmfSender (one in each
 // direction).
 TEST_F(PeerConnectionIntegrationTest, DtmfSenderObserver) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   // Only need audio for DTMF.
   caller()->AddAudioOnlyMediaStream();
   callee()->AddAudioOnlyMediaStream();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  // DTLS must finish before the DTMF sender can be used reliably.
+  ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
   TestDtmfFromSenderToReceiver(caller(), callee());
   TestDtmfFromSenderToReceiver(callee(), caller());
 }
 
 // Basic end-to-end test, verifying media can be encoded/transmitted/decoded
 // between two connections, using DTLS-SRTP.
 TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithDtls) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   // Do normal offer/answer and wait for some frames to be received in each
@@ skipping 1516 matching lines @@
   virtual_socket_server()->set_delay_mean(media_hop_delay_ms);
   virtual_socket_server()->UpdateDelayDistribution();
 
   // Set "offer to receive audio/video" without adding any tracks, so we just
   // set up ICE/DTLS with no media.
   PeerConnectionInterface::RTCOfferAnswerOptions options;
   options.offer_to_receive_audio = 1;
   options.offer_to_receive_video = 1;
   caller()->SetOfferAnswerOptions(options);
   caller()->CreateAndSetAndSignalOffer();
-  // TODO(deadbeef): kIceConnectionConnected currently means both ICE and DTLS
-  // are connected. This is an important distinction. Once we have separate ICE
-  // and DTLS state, this check needs to use the DTLS state.
-  EXPECT_TRUE_SIMULATED_WAIT(
-      (callee()->ice_connection_state() ==
-           webrtc::PeerConnectionInterface::kIceConnectionConnected ||
-       callee()->ice_connection_state() ==
-           webrtc::PeerConnectionInterface::kIceConnectionCompleted) &&
-          (caller()->ice_connection_state() ==
-               webrtc::PeerConnectionInterface::kIceConnectionConnected ||
-           caller()->ice_connection_state() ==
-               webrtc::PeerConnectionInterface::kIceConnectionCompleted),
-      total_connection_time_ms, fake_clock);
+  EXPECT_TRUE_SIMULATED_WAIT(DtlsConnected(), total_connection_time_ms,
+                             fake_clock);
   // Need to free the clients here since they're using things we created on
   // the stack.
   delete SetCallerPcWrapperAndReturnCurrent(nullptr);
   delete SetCalleePcWrapperAndReturnCurrent(nullptr);
 }
 
 // Test that audio and video flow end-to-end when codec names don't use the
 // expected casing, given that they're supposed to be case insensitive. To test
 // this, all but one codec is removed from each media description, and its
 // casing is changed.
@@ skipping 93 matching lines @@
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // Wait for additional audio frames to be received by the callee.
   ExpectNewFramesReceivedWithWait(0, 0, kDefaultExpectedAudioFrameCount, 0,
                                   kMaxWaitForFramesMs);
 }
 
 }  // namespace
 
 #endif  // if !defined(THREAD_SANITIZER)